BUG=2133 R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/39559004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
126 lines
4.5 KiB
C++
Executable File
126 lines
4.5 KiB
C++
Executable File
/*
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* libjingle
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* Copyright 2014 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#ifndef TALK_MEDIA_WEBRTC_SIMULCAST_H_
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#define TALK_MEDIA_WEBRTC_SIMULCAST_H_
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#include <vector>
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#include "webrtc/base/basictypes.h"
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#include "webrtc/config.h"
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namespace webrtc {
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struct VideoCodec;
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}
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namespace cricket {
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struct VideoOptions;
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struct StreamParams;
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enum SimulcastBitrateMode {
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SBM_NORMAL = 0,
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SBM_HIGH,
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SBM_VERY_HIGH,
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SBM_COUNT
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};
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// Config for use with screen cast when temporal layers are enabled.
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struct ScreenshareLayerConfig {
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public:
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ScreenshareLayerConfig(int tl0_bitrate, int tl1_bitrate);
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// Bitrates, for temporal layers 0 and 1.
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int tl0_bitrate_kbps;
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int tl1_bitrate_kbps;
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static ScreenshareLayerConfig GetDefault();
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// Parse bitrate from group name on format "(tl0_bitrate)-(tl1_bitrate)",
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// eg. "100-1000" for the default rates.
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static bool FromFieldTrialGroup(const std::string& group,
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ScreenshareLayerConfig* config);
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};
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// TODO(pthatcher): Write unit tests just for these functions,
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// independent of WebrtcVideoEngine.
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// Get the simulcast bitrate mode to use based on
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// options.video_highest_bitrate.
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SimulcastBitrateMode GetSimulcastBitrateMode(
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const VideoOptions& options);
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// Get the ssrcs of the SIM group from the stream params.
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void GetSimulcastSsrcs(const StreamParams& sp, std::vector<uint32>* ssrcs);
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// Get simulcast settings.
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std::vector<webrtc::VideoStream> GetSimulcastConfig(
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size_t max_streams,
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SimulcastBitrateMode bitrate_mode,
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int width,
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int height,
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int max_bitrate_bps,
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int max_qp,
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int max_framerate);
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// Set the codec->simulcastStreams, codec->width, and codec->height
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// based on the number of ssrcs to use and the bitrate mode to use.
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bool ConfigureSimulcastCodec(int number_ssrcs,
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SimulcastBitrateMode bitrate_mode,
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webrtc::VideoCodec* codec);
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// Set the codec->simulcastStreams, codec->width, and codec->height
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// based on the video options (to get the simulcast bitrate mode) and
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// the stream params (to get the number of ssrcs). This is really a
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// convenience function.
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bool ConfigureSimulcastCodec(const StreamParams& sp,
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const VideoOptions& options,
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webrtc::VideoCodec* codec);
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// Set the numberOfTemporalLayers in each codec->simulcastStreams[i].
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// Apparently it is useful to do this at a different time than
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// ConfigureSimulcastCodec.
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// TODO(pthatcher): Figure out why and put this code into
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// ConfigureSimulcastCodec.
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void ConfigureSimulcastTemporalLayers(
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int num_temporal_layers, webrtc::VideoCodec* codec);
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// Turn off all simulcasting for the given codec.
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void DisableSimulcastCodec(webrtc::VideoCodec* codec);
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// Log useful info about each of the simulcast substreams of the
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// codec.
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void LogSimulcastSubstreams(const webrtc::VideoCodec& codec);
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// Configure the codec's bitrate and temporal layers so that it's good
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// for a screencast in conference mode. Technically, this shouldn't
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// go in simulcast.cc. But it's closely related.
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void ConfigureConferenceModeScreencastCodec(webrtc::VideoCodec* codec);
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} // namespace cricket
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#endif // TALK_MEDIA_WEBRTC_SIMULCAST_H_
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